Method and apparatus for efficiently accounting for the temporal nature of audio processing

ABSTRACT

Some embodiments of the invention provide a computer system for processing an audio track. This system includes at least one DSP for processing the audio track. It also includes an application for editing the audio track. To process audio data in a first interval of the audio track, the application first asks and obtains from the DSP an impulse response parameter related to the DSP&#39;s processing of audio data. From the received impulse response parameter, the application identifies a second audio-track interval that is before the first interval. To process audio data in the first interval, the application then directs the DSP to process audio data within the first and second intervals.

FIELD OF THE INVENTION

The present invention is directed towards method and apparatus forefficiently accounting for the temporal nature of audio processing.

BACKGROUND OF THE INVENTION

Audio processing applications often apply digital signal processing(“DSP”) operations that intentionally modify the audio content of anaudio track. These operations typically cause audio events in the audiodata to have an effect in the audio presentation for an extended periodof time. In other words, certain DSP operations can cause an audio eventto leave a trailing sound effect in the audio presentation even afterthe event finishes. Such a sound effect affects the audio presentationin the absence of a subsequent audio event. It also affects the soundgenerated during a subsequent audio event. Accordingly, audio processingapplications need to account for the temporal effects that can resultfrom applying certain signal processing operations on audio data. Toaccount for such temporal effects on audio data that is within aparticular interval of a track, audio processing applications need toconsider audio data before and/or after the particular interval.

Audio processing applications also re-encode audio data. Re-encodingaudio data might entail re-sampling the audio data, reducing the numberof audio samples, increasing the number of audio samples, changing theencoding format for the audio samples, etc. When such applicationsre-encode an interval of an audio track, they often need to account forcertain number of samples before and after the interval, because of thetemporal nature of audio data.

Accordingly, in a variety of contexts, audio processing applicationsneed to account for the effects of audio data that is before and/orafter a particular segment of audio data that the applications areprocessing. For such contexts, there is a need in the art for a methodthat efficiently accounts for the temporal nature of audio processing.

SUMMARY OF THE INVENTION

Some embodiments of the invention provide a computer system forprocessing an audio track. This system includes at least one DSP forprocessing the audio track. It also includes an application for theediting the audio track. To process audio data in a first interval ofthe audio track, the application first asks and obtains from the DSP animpulse response parameter related to the DSP's processing of audiodata. From the received impulse response parameter, the applicationidentifies a second audio-track interval that is before the firstinterval. To process audio data in the first interval, the applicationthen directs the DSP to process audio data within the first and secondintervals.

BRIEF DESCRIPTION OF THE DRAWINGS

The novel features of the invention are set forth in the appendedclaims. However, for purpose of explanation, several embodiments of theinvention are set forth in the following figures.

FIG. 1 illustrates the software architecture of a computer system thatis used in conjunction with some embodiments of the invention.

FIGS. 2-5 present different illustrations of an audio track in order todescribe the need to account for temporal audio effects.

FIG. 6 illustrates on prior art process that renders audio data in aparticular interval of an audio track.

FIG. 7 illustrates a process that accounts for temporal effects on audiowhile processing the audio in an audio unit that is designed to modifythe content of the audio.

FIG. 8 illustrates an encoding process that in encoding an interval inan audio track examines samples before and after the interval.

DETAILED DESCRIPTION OF THE INVENTION

In the following description, numerous details are set forth for purposeof explanation. However, one of ordinary skill in the art will realizethat the invention may be practiced without the use of these specificdetails. In other instances, well-known structures and devices are shownin block diagram form in order not to obscure the description of theinvention with unnecessary detail.

I. Audio Processing System

FIG. 1 illustrates the software architecture of a computer system 100that is used in conjunction with some embodiments of the invention. Asshown in this figure, this computer system includes several audio tracks105, several audio units 110, several audio converters 115, and anaudio-editing application 120. A user of the computer system 100 caninteract with the audio-editing application 120 to edit and combineaudio tracks 105 in order to make an audio presentation.

To create an audio presentation, the audio-editing application 120 mightdirect one or more audio units 110 to modify the audio content of one ormore audio tracks by performing a set of DSP operations on the audiocontent. In other words, each audio unit 110 performs a set of DSPoperations on audio data that it receives from the audio application, inorder to change this audio data.

To create an audio presentation, the audio-editing application 120 mightalso direct one or more audio converters 115 to format the audio data.Each audio converter 115 performs a particular set of DSP operations onthe audio data that it receives from the application 120, in order toencode or change the encoding of the audio data. Examples of differentencoding operations include re-sampling the audio data, reducing thenumber of audio samples, increasing the number of audio samples,translating the audio encoding from one standard (e.g., an mp3 format)to another standard (e.g., linear PCM format). The encoding of the audiodata might also change the audio data, as some encodings (such as mp3)are lossy. However, the difference between an audio converter and anaudio unit is that the operations of the audio unit are designed tochange the received audio content, while the operations of the audioconverter are designed to keep the received audio content as close tothe original content as possible. The resulting change due to an audioconverter's operation is typically undesirable and unavoidable.

As shown in FIG. 1, each audio unit 110 or converter 115 includes a DSP125, which might be formed by one individual digital signal processor orby several individual digital signal processors. An audio unit 110 usesits DSP to perform its content-modifying operations, while a converter115 uses its DSP to perform its encoding operation. To use an audio unit110 or a converter 115 to process audio data in a first interval of theaudio track, the audio-editing application 120 first asks and obtainsfrom the audio unit or converter a duration parameter related to itsDSP's processing of audio data. From the received duration parameter,the application 120 identifies a second audio-track interval that isbefore the first interval. To process audio data in the first interval,the application then directs the DSP to process audio data within thefirst and second time intervals.

In some embodiments, the duration parameter specifies the duration ofthe impulse response of the DSP. In case when the DSP is part of anaudio converter 115 that re-encodes the audio data, the durationparameter is called a priming duration parameter. A priming durationparameter specifies the amount of audio data that the audio conversionoperation needs to consider before or after a particular interval inorder to process audio data within the particular time interval. Apre-priming parameter specifies the duration of audio data to considerbefore the particular interval, while a post-priming parameter specifiesthe duration of audio data to consider after the particular interval.

The duration parameter can be expressed differently in differentsituations. For instance, it can be expressed in terms of time (i.e., itcan directly express the duration of a time interval, e.g., 0.5seconds). It also can be expressed in terms of a number of samples(e.g., it can specify 10 samples). The sample count is converted into atime interval in some cases, while it is directly used in other cases.

II. Interaction Between the Audio-Editing Application and an Audio Unit

FIG. 7 presents a process 700 that conceptually illustrates what isperformed by the audio-editing application 120 and an audio unit 110 inorder to process audio data within a particular time interval. Asfurther described below, this process considers audio data before andafter the particular interval in order to process the audio data duringthe particular interval. Before describing this process, however, theneed to account for audio data outside of the particular interval isdescribed by reference to FIGS. 2-5.

FIG. 2 illustrates an audio track 200 with four audio events 205, 210,215, and 220 that occur at four different times, t1, t2, t3, and t4, inthe track. Each of the four audio events is an impulse audio signal. InFIG. 2, no DSP effect has been applied to any of the audio events.Hence, none of the audio events results in an audio signal that lastsbeyond its duration. In other words, the audio contribution of eachevent terminates once each event terminates.

FIG. 3 presents a graph of the audio track 200 after an audio unit hasapplied a reverb effect to this track. As shown in this figure, thiseffect generates a reverb audio signal 305 for each audio event. Eachreverb signal trails its audio event and decays after its event. In thisexample, the reverb 305 a of the first event 205 overlaps with thesecond event 210, the reverb 305 b of the second event 210 overlaps withthe third event 215, and the reverb 305 c of the third event 215overlaps with the fourth event 220. The overlap of a reverb of aninitial event with a subsequent event and the subsequent event's reverbmodifies the sound that is generated during the subsequent event and thesound that is generated after the subsequent event. FIGS. 3-5 do notshow the modification of a subsequent event's reverb due to a previousevent's reverb, in order to keep the visual presentation of theseexamples simple.

FIG. 4 illustrates a graph of the audio track 200 when this track isplayed from a time t5 to a time t6. This figure illustrates a playhead405 that is initially positioned at the time t5 on the horizontal timeaxis of the graph. When the audio track is played, this playhead scrollsacross the horizontal time axis to indicate the position in the audiotrack that is being played at any instant in time. In the exampleillustrated in FIG. 4, the audio processing applications ignore thereverb signal 305 a of the audio event 205 that occurs at time t1, whichis before the starting time t5 of the playhead. Hence, in this example,the audio that is played starting at time t5 does not accuratelyrepresent the application of the reverb effect on the audio track, as itdoes not account for audio contributions from audio event 205 before,during, and after the event 210.

Such inaccurate representation can be troublesome for a variety ofreasons. For instance, an inaccurate audio representation makes breakingan audio production in different sections that are stored on differentmedia difficult. The missing DSP effects at the start of a tape willcreate an audible discontinuity when switching to the tape from anothertape.

FIG. 5 illustrates a graph of the audio track 200 when this track isplayed from a time t5 to a time t6. Like FIG. 4, this figure illustratesa playhead 405 that starts on the horizontal time axis at time t5 andscrolls to time t6 while the audio track is playing. However, unlike theexample illustrated in FIG. 4, the audio track in FIG. 5 has beenprocessed to account for the reverb signal of the audio event 205 thatoccurs at time t1, which is before the starting time t5 of the playhead.Specifically, in this example, the audio track has been processed toinclude the reverb contribution 505 after time t5 of the reverb signal305 a. Hence, in this example, the audio that is played starting at timet5 accurately represents the application of the reverb effect on theaudio track, as it accounts for audio contributions 505 from audio event205 before, during, and after the event 210.

FIG. 6 illustrates one prior art process 600 that renders audio data ina particular interval of an audio track. The interval starts at a timet1 and ends at a time t2. This process accounts for audio data beforeand/or after the particular interval in order to account for temporaleffects on audio data that is within the particular interval. Thisprocess is performed by an audio-editing application and a DSP.

As shown in FIG. 6, the process 600 starts (at 605) by the audio-editingapplication informing the DSP that it wants to process audio from timesn to m. In the first pass through 605, the process specifies n to equalthe start time t1, and m to be t1 plus some fraction of the differencebetween the start and end times t1 and t2 of the interval.

Next, at 610, the DSP asks the application for audio samples from timen−p to time m, where p is a DSP-computed value that specifies the priorduration of samples that it needs to examine in order to accuratelyprocess samples from times n to m. At 615, the application then providesthe DSP with the requested samples from times n−p to m. At 620, the DSPthen processes the audio and provides the application with processedaudio data from times n to m. The application then outputs processedaudio data from times n to m.

The application then determines (at 625) whether the variable m equalsthe end time t2. If not, the application (at 630) sets n to m, and setsm to n plus the difference between the previous n and m. After 630, theprocess performs 605 and its subsequent operations, which were describedabove.

When the application determines (at 630) that the variable m equals theend time t2, it directs (at 640) the DSP to process audio data aftertime m for a set amount of time and outputs this processed audio. After640, the process ends. The process illustrated in FIG. 6 is inefficientas, each time the DSP is processing a set of samples, it has to ask forsamples before this set. It also requires the DSP to identify and useits duration parameter each time.

FIG. 7 illustrates a process 700 of some embodiments of the invention.This process renders audio data in a particular interval of an audiotrack. The interval starts at a time t1 and ends at a time t2. Thisprocess accounts for audio data before and/or after the particularinterval in order to account for temporal effects on audio data that iswithin the particular interval. This process is performed by anaudio-editing application and a DSP of an audio unit.

As shown in FIG. 7, the process 700 starts (at 705) when theaudio-editing application 120 asks a DSP 125 for the duration of theDSP's impulse response time. This duration is called the effectsduration or tail time below. In response, the DSP provides (at 710) itstail time T to the application. The application then asks (at 715) theDSP for its latency duration parameter. This parameter specifies theduration of time that the DSP takes after receiving an audio signal tooutput a signal related to the received signal. The DSP supplies (at720) its latency duration parameter L. In the embodiment illustrated inFIG. 7, both the effect and latency duration parameters are expressed inunits of time (e.g., they specify 0.5 and 0.2 seconds). In otherembodiments, these parameters might be expressed in terms of the numberof samples.

Next, at 725, the application defines three variables n, m, and p.Specifically, at 725, the application (1) defines n to be equal to thestart time t1, (2) define m to be equal to t1 plus a delta, where thedelta is typically much smaller than the difference between the startand end times t1 and t2, and (3) defines p to be equal to T+L. At 725,the application then asks the DSP to process audio samples from timesn−p to m.

In response, the DSP processes (at 730) the audio samples from times n−pto n. The nature of this processing depends on the DSP and the DSP'saudio unit. Also, processing audio samples is well known in the art, asthere are a variety of commonly known techniques for such processing.See, e.g., Digital Audio Signal Processing by Udo Zoizer, published byJohn Wiley & Son Ltd; (August 1997). In order not to obscure thedescription of the invention with unnecessary detail, the processing ofaudio samples by a DSP will not be further described below. At 730, theDSP provides to the application processed audio samples for times n−p ton. In some embodiments, the application discards these samples.

Next, at 735, the audio-editing application determines whether thevariable m equals time t2. If not, the application asks (at 740) the DSPto process audio samples from times n to m. In response, the DSPprocesses (at 745) the audio samples from times n to m. At 745, the DSPalso provides the processed audio samples for times n to m to the audioediting application. This application then outputs (at 750) theprocessed audio samples for times n to m. Outputting the processed audiosamples might entail (1) providing an audio presentation to a user basedon the processed samples, (2) storing the audio samples, or (3) havinganother DSP process these samples.

After 750, the audio-editing application then sets n to m, and sets m ton plus the difference between the previous n and m (e.g., if n and mrespectively were 2 and 2.1, the application will set n to 2.1 and m to2.2). From 755, the process transitions back to 735, which was describedabove.

When the audio-editing application determines (at 735) that m equalstime t2, the application has the DSP perform a post-interval processingthat is meant to capture properly the temporal effects of samples withinthe interval from t1 to t2, on samples outside of this interval.Specifically, the application asks (at 760) the DSP to process audiosamples from times m to m+p. In response, the DSP processes (at 765) theaudio samples from times m to m+p. At 770, the DSP also provides theprocessed audio samples for times m to m+p to the audio editingapplication. This application then outputs (at 770) the processed audiosamples for times m to m+p. The process 700 then ends.

Although the process 700 is described above in one manner, one ofordinary skill will realize that other embodiments might implement thisprocess differently. For instance, in other embodiments, the applicationprovides (at 760) the DSP with zero samples. In this manner, theapplication pushes silence through the DSP in order to only receive thetrailing audio effect of samples with the interval from t1 to t2, andnot involve samples from t2 to t2+p.

The inventive process 700 of FIG. 7 has several advantages. First, itaccurately performs audio tail editing, which is important in a varietyof contexts. For instance, video editing applications typically break upa movie in two or more different parts and they output each part on adifferent tape. When this occurs, the audio at the beginning of eachtape after the first tape needs to account for trailing effects of theaudio at the end of the previous tape. Otherwise, the viewer willdiscern an audio distortion as the presentation transitions from onetape to another. Second, the process 700 has a DSP publish its tail timeand its latency to the editing application. This reduces the amount ofprocessing that the DSP has to perform. It also allows the pre- andpost-processing to be more accurate as, for each DSP, this processing isdependent on the DSP's own unique tail and latency times.

Although the process 700 was described above by reference to one DSP,one of ordinary skill will realize that the audio-editing applicationcan perform this process concurrently for several DSPs. For instance,once one DSP finishes processing a set of samples from times n to m, theaudio editing application can supply the processed samples to anotherDSP for processing.

III. Interaction Between the Audio-Editing Application and an AudioConverter

As mentioned above, an audio converter 115 performs a particular set ofencoding operations on the audio data that it receives from theapplication 120. Examples of different encoding operations includere-sampling the audio data, reducing the number of audio samples,increasing the number of audio samples, translating the audio encodingfrom one standard (e.g., an mp3 format) to another standard (e.g.,linear PCM format).

To encode audio samples in an interval between time t1 and t2, encodingoperations often need to examine samples before time t1 and samplesafter time t2. FIG. 8 illustrates one such encoding process 800 of someembodiments of the invention. The audio-editing application and a DSP ofan audio converter perform this process.

As shown in FIG. 8, the process 800 starts (at 805) when theaudio-editing application 120 asks a DSP 125 for the DSP's pre-primingduration parameter. As mentioned above, a pre-priming parameterspecifies the duration of audio data to consider before the intervalbeing considered, while a post-priming parameter specifies the durationof audio data to consider after the interval. In some embodiments, theinterval is expressed in terms of a temporal duration. In the embodimentillustrated in FIG. 8, the interval is a particular set of samples inthe audio track (e.g., samples 500 to 550).

In response to the request, the DSP provides (at 810) its pre-primingparameter p to the audio-editing application. The application then asks(at 815) the DSP for its post-priming parameter. The DSP supplies (at820) its post-priming parameter x. In the embodiment illustrated in FIG.8, the pre- and post-priming parameters each specify a number of samples(e.g., 5 and 7 samples). In other embodiments, they can be expressed interms of a time value (e.g., 0.5 and 0.2 seconds).

Next, at 825, the application defines two variables n and m.Specifically, at 825, the application (1) defines n to be equal to theStart parameter that specifies the start of the interval, and (2) definem to be equal to n plus a delta, where the delta is typically muchsmaller than the difference between the Start and End parameters thatspecify the duration of the interval. At 825, the application then asksthe DSP to process audio samples in the interval n−p to m.

In response, the DSP processes (at 830) the audio samples in theinterval n−p to m. The nature of this processing depends on the DSP andthe DSP's audio converter. Also, processing audio samples in an audioconverter is well known in the art, as there are a variety of commonlyknown techniques for such processing. See, e.g., Digital Audio SignalProcessing by Udo Zolzer, published by John Wiley & Son Ltd; (August1997). In order not to obscure the description of the invention withunnecessary detail, the processing of audio samples by an audioconverter's DSP will not be further described below. After processingthe samples, the DSP provides (at 830) samples for the interval n to m−pto the audio-editing application.

The audio-editing application then outputs (at 835) the received samplesfor the interval n to m−p. Outputting the processed audio samples mightentail (1) providing an audio presentation to a user based on theprocessed samples, (2) storing the audio samples, or (3) having anotherDSP process these samples.

Next, at 840, the application determines whether the variable m equalsthe End parameter that specifies the end of the interval. If not, theapplication (at 845) sets n to m, and sets m to new n plus thedifference between the previous m and the previous n (e.g., if n and mrespectively were 2 and 2.1, the application will set n to 2.1 and m to2.2). The application then asks (at 850) the DSP to process audiosamples from times n to m. In response, the DSP processes (at 855) theaudio samples from times n−p to m−p. At 855, the DSP also provides theprocessed audio samples for times n−p to m−p to the audio editingapplication. This application then outputs (at 860) the processed audiosamples for times n−p to m−p. From 860, the process transitions back to840, which was described above.

When the application determines (at 840) that m equals End, theapplication directs the DSP to perform a post-processing that is neededto obtain the samples from m−p to m, so that it can complete itsprocessing of the samples that were originally in the interval definedby Start and End in the interval n to m. Specifically, at 865, theapplication sets n to m, and set m to the new n plus the post-primingparameter x. The application also asks (at 865) the DSP to process audiosamples from times n to m. In response, the DSP processes (at 870) theseaudio samples, and provides to the application audio samples for timesn−p to n. At 875, the application then outputs the received processedaudio samples and then ends the operation of the process 800.

While the invention has been described with reference to numerousspecific details, one of ordinary skill in the art will recognize thatthe invention can be embodied in other specific forms without departingfrom the spirit of the invention. Thus, one of ordinary skill in the artwould understand that the invention is not to be limited by theforegoing illustrative details, but rather is to be defined by theappended claims.

1. An audio processing system comprising: a) an audio track comprisingaudio data; b) a digital signal processor (“DSP”) for processing theaudio track; c) an application for editing the audio track, wherein, toprocess audio data in a first interval of the audio track, theapplication (i) requests from the DSP an impulse response parameterrelated to the DSP's impulse response, (ii) receives from the DSP theimpulse response parameter, (iii) from the impulse response parametercomputes a second interval for specifying an interval of audio dataneeded by the DSP to account for temporal effects affecting the audiodata in the first interval, and (iv) directs the DSP to process audiodata within the first and second intervals, wherein the applicationreuses said impulse response parameter to account for temporal effectsaffecting subsequently processed audio data.
 2. The audio processingsystem of claim 1, wherein the DSP is part of an audio unit that isdesigned to modify the audio data so that the audio data can produce adifferent sound.
 3. The audio processing system of claim 1, wherein theDSP is a single digital signal processor.
 4. The audio processing systemof claim 1, wherein the DSP is formed by several digital signalprocessors.
 5. The audio processing system of claim 1, wherein theapplication also asks the DSP for a latency parameter of the DSP.
 6. Theaudio processing system of claim 1, wherein the DSP is part of an audioconverter that re-encodes the audio data.
 7. The audio processing systemof claim 6, wherein the impulse response parameter is a priming durationparameter that specifies an amount of audio data that the audioconverter needs to consider in order to process audio data within thefirst interval.
 8. The audio processing system of claim 7, wherein thepriming parameter is a pre-priming parameter that specifies a durationof audio data to consider before the first interval.
 9. The audioprocessing system of claim 8, wherein the priming parameter alsoincludes a post-priming parameter that specifies a duration of audiodata to consider after the first interval.
 10. The audio processingsystem of claim 7, wherein the priming parameter is a post-primingparameter that specifies a duration of audio data to consider after thefirst interval.
 11. A method for processing audio data in a firstinterval of an audio track, said method for use with an application thatedits the audio data of the audio track, the method comprising:receiving an impulse response parameter of a particular digital signalprocessor (“DSP”)from the particular DSP, computing, at the application,a second interval based on the impulse response parameter to account fortemporal effects affecting the audio data in the first interval, anddirecting the particular DSP to process audio data within the first andsecond intervals, wherein said application reuses the received durationparameter to account for temporal effects affecting subsequentlyprocessed audio data.
 12. The audio processing method of claim 11,wherein the DSP is part of an audio converter that re-encodes the audiodata.
 13. The audio processing method of claim 12, wherein the impulseresponse parameter is a priming duration parameter that specifies anamount of audio data that the audio converter needs to consider in orderto process audio data within the first interval.
 14. The audioprocessing method of claim 13, wherein the priming parameter is apre-priming parameter that specifies a duration of audio data toconsider before the first interval.
 15. The audio processing method ofclaim 14, wherein the priming parameter also includes a post-primingparameter that specifies a duration of audio data to consider after thefirst interval.
 16. The audio processing method of claim 13, wherein thepriming parameter is a post-priming parameter that specifies a durationof audio data to consider after the first interval.
 17. The audioprocessing system of claim 7, wherein the amount of audio data isexpressed in a number of samples.
 18. A system comprising: a) an audioprocessor for processing audio data; b) an application for editing audiodata of an audio track, wherein to process audio data in an interval,said application (i) receives from the audio processor an operationalparameter of the audio processor, (ii) generates a duration from theoperational parameter, (iii) identifies audio data outside the intervalusing the duration, wherein said audio data outside the interval is foraccurately processing temporal effects affecting the audio data in theinterval and (iv) directs the audio processor to process audio databetween the interval and the audio data outside the interval.
 19. Thesystem of claim 18, wherein the duration is expressed in a number ofaudio samples.
 20. The system of claim 18, wherein the operationalparameter comprises an impulse response parameter of the audioprocessor.
 21. The system of claim 18, wherein the operational parametercomprises a latency parameter of the audio processor.
 22. The system ofclaim 18, wherein the duration specifies an amount of audio data thatthe audio processor needs to consider before the interval in order toprocess audio data within the interval.
 23. The system of claim 22,wherein the duration further specifies an amount of audio data that theaudio processor needs to consider after the interval in order to processaudio data within the interval.
 24. The system of claim 18, wherein theduration specifies an amount of audio data that the audio processorneeds to consider after the interval in order to process audio datawithin the interval.
 25. A computer-readable medium storing anapplication which when executed by at least one processor permits forthe editing of audio data within an interval of an audio track, whereinthe application uses an audio processor for processing the audio data,the application comprising sets of instructions for: receiving aduration parameter of the audio processor from the audio processor;identifying audio data outside the interval based on the durationparameter, wherein said audio data outside the interval is foraccurately processing temporal effects affecting the audio data in theinterval; and directing the audio processor to process audio databetween the interval and the identified audio data outside the interval,wherein the application reuses the received duration parameter toaccount for temporal effects affecting subsequently processed audiodata.
 26. The computer-readable medium of claim 25, wherein the durationparameter specifies an amount of audio data that the audio processorneeds to consider before the interval in order to process audio datawithin the interval.
 27. The computer-readable medium of claim 25,wherein the duration parameter specifies an amount of audio data thatthe audio processor needs to consider after the interval in order toprocess audio data within the interval.
 28. The computer-readable mediumof claim 25, wherein the duration parameter comprises an impulseresponse parameter of the audio processor.
 29. The computer-readablemedium of claim 25, wherein the duration parameter comprises a latencyparameter of the audio processor.
 30. A method for processing audio datain an interval of an audio track, said method performed by anapplication that edits the audio data of the audio track, the methodcomprising: receiving, at the application, an operational parameter of aparticular audio processor from the particular audio processor;generating a duration from the operational parameter; identifying audiodata outside the interval based on the duration, wherein said audio dataoutside the interval is for accurately processing temporal effectsaffecting the audio data in the interval; and directing the particularaudio processor to process audio data between the interval and theidentified audio data outside the interval.